Audio conferencing method

ABSTRACT

An audio conferencing system comprises an audio conference mixer that receives digitized audio signals and sums a plurality of the digitized audio signals containing speech to provide a summed conference signal. A transcoder receives and transcodes the summed conference signal to provide a transcoded summed signal that is streamed onto the Internet.

CROSS REFERENCE TO RELATED APPLICATIONS

[0001] This application contains subject matter related to a commonlyassigned co-pending application designated Ser. No. TBD, filed Mar. 22,2000, entitled “Scalable Audio Conference Platform”. This application ishereby incorporated herein by reference.

BACKGROUND OF THE INVENTION

[0002] The present invention relates to telephony, and in particular toan audio conferencing platform.

[0003] Audio conferencing platforms are well known. For example, seeU.S. Pat. Nos. 5,483,588 and 5,495,522. Audio conferencing platformsallow conference participants to easily schedule and conduct audioconferences with a large number of users. In addition, audio conferenceplatforms are generally capable of simultaneously supporting manyconferences.

[0004] Due to the widespread popularity of the World Wide Web, Internettraffic is at an all time high and rapidly increasing. In addition, themove towards IP communications is gathering momentum. Users arecurrently using the Internet as a mechanism for retrieving streamedaudio and video media streams.

[0005] There is a need for an audio conferencing system that can streamits summed conference audio onto the Internet in real-time. This willallow a user to listen to an audio conference supported by the audioconferencing system, over the Internet.

SUMMARY OF THE INVENTION

[0006] Briefly, according to the present invention, an audioconferencing system comprises an audio conference mixer that receivesdigitized audio signals and sums a plurality of the digitized audiosignals containing speech to provide a summed conference signal. Atranscoder receives and transcodes the summed conference signal toprovide a transcoded summed signal that is streamed onto the Internet.

[0007] In one embodiment an audio conferencing platform includes a databus, a controller, and an interface circuit that receives audio signalsfrom a plurality of conference participants and provides digitized audiosignals in assigned time slots over the data bus. The audio conferencingplatform also includes a plurality of digital signal processors (DSPs)adapted to communicate on the TDM bus with the interface circuit. Atleast one of the DSPs sums a plurality of the digitized audio signalsassociated with conference participants who are speaking to provide asummed conference signal. This DSP provides the summed conference signalto at least one of the other plurality of DSPs, which removes thedigitized audio signal associated with a speaker whose voice is includedin the summed conference signal, thus providing a customized conferenceaudio signal to each of the speakers.

[0008] In a preferred embodiment, the audio conferencing platformconfigures at least one of the DSPs as a centralized audio mixer and atleast another one of the DSPs as an audio processor. Significantly, thecentralized audio mixer performs the step of summing a plurality of thedigitized audio signals associated with conference participants who arespeaking, to provide the summed conference signal. The centralized audiomixer provides the summed conference signal to the audio processor(s)for post processing and routing to the conference participants. The postprocessing includes removing the audio associated with a speaker fromthe conference signal to be sent to the speaker. For example, if thereare forty conference participants and three of the participants arespeaking, then the summed conference signal will include the audio fromthe three speakers. The summed conference signal is made available onthe data bus to the thirty-seven non-speaking conference participants.However, the three speakers each receive an audio signal that is equalto the summed conference signal less the digitized audio signalassociated with the speaker. Removing the speaker's voice from the audiohe hears reduces echoes.

[0009] The centralized audio mixer also receives DTMF detect bitsindicative of the digitized audio signals that include a DTMF tone. TheDTMF detect bits may be provided by another of the DSPs that isprogrammed to detect DTMF tones. If the digitized audio signal isassociated with a speaker, but the digitized audio signal includes aDTMF tone, the centralized conference mixer will not include thedigitized audio signal in the summed conference signal while that DTMFdetect bit signal is active. This ensures conference participants do nothear annoying DTMF tones in the conference audio. When the DTMF tone isno longer present in the digitized audio signal, the centralizedconference mixer may include the audio signal in the summed conferencesignal.

[0010] The audio conference platform is capable of supporting a numberof simultaneous conferences (e.g., 384). As a result, the audioconference mixer provides a summed conference signal for each of theconferences.

[0011] Each of the digitized audio signals may be preprocessed. Thepreprocessing steps include decompressing the signal (e.g., μ-Law orA-Law compression), and determining if the magnitude of the decompressedaudio signal is greater than a detection threshold. If it is, then aspeech bit associated with the digitized audio signal is set. Otherwise,the speech bit is cleared.

[0012] These and other objects, features and advantages of the presentinvention will become apparent in light of the following detaileddescription of preferred embodiments thereof, as illustrated in theaccompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

[0013]FIG. 1 is a pictorial illustration of a conferencing system;

[0014]FIG. 2 illustrates a functional block diagram of an audioconferencing platform within the conferencing system of FIG. 1;

[0015]FIG. 3 is a block diagram illustration of a processor board withinthe audio conferencing platform of FIG. 2;

[0016]FIG. 4 is a functional block diagram illustration of the resourceson the processor board of FIG. 3;

[0017]FIG. 5 is a flow chart illustration of audio processor processingfor signals received from the network interface cards over the TDM bus;

[0018]FIG. 6 is a flow chart illustration of the DTMF tone detectionprocessing;

[0019] FIGS. 7A-7B together provide a flow chart illustration of theconference mixer processing to create a summed conference signal;

[0020]FIG. 8 is a flow chart illustration of audio processor processingfor signals to be output to the network interface cards via the TDM bus;and

[0021]FIG. 9 is a flow chart illustration of the transcoding performedon the summed conference signal(s) to provide “real-time” conferenceaudio over the Internet.

DETAILED DESCRIPTION OF THE INVENTION

[0022]FIG. 1 is a pictorial illustration of a conferencing system 20.The system 20 connects a plurality of user sites 21-23 through aswitching network 24 to an audio conferencing platform 26. The pluralityof user sites may be distributed worldwide, or at a companyfacility/campus. For example, each of the user sites 21-23 may be indifferent cities and connected to the audio platform 26 via theswitching network 24, that may include PSTN and PBX systems. Theconnections between the user sites and the switching network 24 mayinclude T1, E1, T3 and ISDN lines.

[0023] Each user site, 21-23 preferably includes a telephone, 28 and acomputer/server 30. However, a conferences site may only include eitherthe telephone or the computer/server. The computer/server 30 may beconnected via an Internet/intranet backbone 32 to a server 34. The audioconferencing platform 26 and the server 34 are connected via a datalink. 36 (e.g., a 10/100 BaseT Ethernet link). The computer 30 allowsthe user to participate in a data conference simultaneous to the audioconference via the server 34. In addition, the user can use the computer30 to interface (e.g., via a browser) with the server 34 to performfunctions such as conference control, administration (e.g., systemconfiguration, billing, reports, . . . ), scheduling and accountmaintenance. The telephone 28 and the computer 30 may cooperate toprovide voice over the Internet/intranet 32 to the audio conferencingplatform 26 via the data link 36.

[0024]FIG. 2 illustrates a functional block diagram of the audioconferencing platform 26. The audio conferencing platform 26 includes aplurality of network interface cards (NICs) 38-40 that receive audioinformation from the switching network 24 (FIG. 1). Each NIC may becapable of handling a plurality of different trunk lines (e.g., eight).The data received by the NIC is generally an 8-bit μ-Law or A-Lawsample. The NIC places the sample into a memory device (not shown),which is used to output the audio data onto a data bus. The data bus ispreferably a time division multiplex (TDM) bus, for example based uponthe H.110 telephony standard.

[0025] The audio conferencing platform 26 also includes a plurality ofprocessor boards 44-46 that receive and transmit data to the NICs 3840over the TDM bus 42. The NICs and the processor boards 44-46 alsocommunicate with a controller/CPU board 48 over a system bus 50. Thesystem bus 50 is preferably based upon the compact PCi standard. TheCPU/controller communicates with the server 34 (FIG. 1) via the datalink 36. The controller/CPU board may include a general purposeprocessor such as a 200 MHz Pentium™ CPU manufactured by IntelCorporation, a processor from AMD or any other similar processor(including an ASIC) having sufficient MIPS to support the presentinvention.

[0026]FIG. 3 is block diagram illustration of the processor board 44 ofthe audio conferencing platform. The board 44 includes a plurality ofdynamically programmable digital signal processors 60-65. Each digitalsignal processor (DSP) is an integrated circuit that communicates withthe controller/CPU card 48 (FIG. 2) over the system bus 50.Specifically, the processor board 44 includes a bus interface 68 thatinterconnects the DSPs 60-65 to the system bus 50. Each DSP alsoincludes an associated dual port RAM (DPR) 70-75 that buffers commandsand data for transmission between the system bus 50 and the associatedDSP.

[0027] Each DSP 60-65 also transmits data over and receives data fromthe TDM bus 42. The processor card 44 includes a TDM bus interface 78that performs any necessary signal conditioning and transformation. Forexample, if the TDM bus is a H.110 bus then it includes thirty-twoserial lines, as a result the TDM bus interface may include aserial-to-parallel and a parallel-to-serial interface. An example, of aserial-to-parallel and a parallel-to-serial interface is disclosed incommonly assigned U.S. Provisional Patent Application designated serialNo. 60/105,369 filed Oct. 23, 1998 and entitled“Serial-to-Parallel/Parallel-to-Serial Conversion Engine”. Thisapplication is hereby incorporated by reference.

[0028] Each DSP 60-65 also includes an associated TDM dual port-RAM80-85 that buffers data for transmission between the TDM bus 42 and theassociated DSP.

[0029] Each of the DSPs is preferably a general purpose digital signalprocessor IC, such as the model number TMS320C6201 processor availablefrom Texas Instruments. The number of DSPs resident on the processorboard 44 is a function of the size of the integrated circuits, theirpower consumption and the heat dissipation ability of the processorboard. For example, there may be between four and ten DSPs per processorboard.

[0030] Executable software applications may be downloaded from thecontroller/CPU 48 (FIG. 2) via the system bus 50 to a selected one(s) ofthe DSPs 60-65. Each of the DSPs is also connected to an adjacent DSPvia a serial data link.

[0031]FIG. 4 is a functional illustration of the DSP resources on theprocessor board 44 illustrated in FIG. 3. Referring to FIGS. 3 and 4,the controller/CPU 48 (FIG. 2) downloads executable program instructionsto a DSP based upon the function that the controller/CPU assigns to theDSP. For example, the controller/CPU may download executable programinstructions for the DSP₃ 62 to function as an audio conference mixer90, while the DSP₂ 61 and the DSP₄ 63 may be configured as audioprocessors 92, 94, respectively. DSP₅ 64 may be configured to performtranscoding 95 on the conference sums in order to provide an audioconference signal suitable for transmission over the Internet inreal-time. This feature will be discussed in detail hereinafter.Significantly, this allows users to listen to the audio conference viathe Internet (i.e., using packet switched audio). Other DSPs 60, 65 maybe configured by the controller/CPU 48 (FIG. 2) to provide services suchas DTMF detection 96, audio message generation 98 and music play back100.

[0032] Each audio processor 92, 94 is capable of supporting a certainnumber of user ports (i.e., conference participants). This number isbased upon the operational speed of the various components within theprocessor board, and the over-all design of the system. Each audioprocessor 92, 94 receives compressed audio data 102 from the conferenceparticipants over the TDM bus 42.

[0033] The TDM bus 42 may support 4096 time slots, each having abandwidth of 64 kbps. The timeslots are generally dynamically assignedby the controller/CPU 48 (FIG. 2) as needed for the conferences that arecurrently occurring. However, one of ordinary skill in the art willrecognize that in a static system the timeslots may be nailed up.

[0034]FIG. 5 is a flow chart illustration of processing steps 500performed by each audio processor on the digitized audio signalsreceived over the TDM bus 42 from the NICs 38-40 (FIG. 2). Theexecutable program instructions associated with these processing steps500 are typically downloaded to the audio processors 92, 94 (FIG. 4) bythe controller/CPU 48 (FIG. 2). The download may occur during systeminitialization or reconfiguration. These processing steps 500 areexecuted at least once every 125 μseconds to provide audio of therequisite quality.

[0035] For each of the active/assigned ports for the audio processor,step 502 reads the audio data for that port from the TDM dual port RAMassociated with the audio processor. For example, if DSP₂ 61 (FIG. 3) isconfigured to perform the function of audio processors 92 (FIG. 4), thenthe data is read from the read bank of the TDM dual port RAM 81. If theaudio processor 92 is responsible for 700 active/assigned ports, thenstep 502 reads the 700 bytes of associated audio data from the TDM dualport RAM 81. Each audio processor includes a time slot allocation table(not shown) that specifies the address location in the TDM dual port RAMfor the audio data from each port.

[0036] Since each of the audio signals is compressed (e.g., μ-Law,A-Law, etc), step 604 decompresses each of the 8-bit signals to a 16-bitword. Step 506 computes the average magnitude (AVM) for each of thedecompressed signals associated with the ports assigned to the audioprocessor.

[0037] Step 508 is performed next to determine which of the ports arespeaking. This step compares the average magnitude for the port computedin step 506 against a predetermined magnitude value representative ofspeech (e.g., −35 dBm). If average magnitude for the port exceeds thepredetermined magnitude value representative of speech, a speech bitassociated with the port is set. Otherwise, the associated speech bit iscleared. Each port has an associated speech bit. Step 510 outputs allthe speech bits (eight per timeslot) onto the TDM bus. Step 512 isperformed to calculate an automatic gain correction (AGC) factor foreach port. To compute an AGC value for the port, the AVM value isconverted to an index value associated with a table containinggain/attenuation factors. For example, there may be 256 index values,each uniquely associated with 256 gain/attenuation factors. The indexvalue is used by the conference mixer 90 (FIG. 4) to determine thegain/attenuation factor to be applied to an audio signal that will besummed to create the conference sum signal.

[0038]FIG. 6 is a flow chart illustration of the DTMF tone detectionprocessing 600. These processing steps. 600 are performed by the DTMFprocessor 96 (FIG. 4), preferably at least once every 125 μseconds, todetect DTMF tones within on the digitized audio signals from the NICs38-40 (FIG. 2). One or more of the DSPs may be configured to operate asa DTMF tone detector. The executable program instructions associatedwith the processing steps 600 are typically downloaded by thecontroller/CPU 48 (FIG. 2) to the DSP designated to perform the DTMFtone detection function. The download may occur during initialization orsystem reconfiguration.

[0039] For an assigned number of the active/assigned ports of theconferencing system, step 602 reads the audio data for the port from theTDM dual port RAM associated with the DSP(s) configured to perform theDTMF tone detection function. Step 604 then expands the 8-bit signal toa 16-bit word. Next, step 606 tests each of these decompressed audiosignals to determine if any of the signals includes a DTMF tone. For anysignal that does include a DTMF tone, step 606 sets a DTMF detect bitassociated with the port. Otherwise, the DTMF detect bit is cleared.Each port has an associated DTMF detect bit. Step 608 informs thecontroller/CPU 48 (FIG. 3) which DTMF tone was detected, since the toneis representative of system commands and/or data from a conferenceparticipant. Step 610 outputs the DTMF detect bits onto the TDM bus.

[0040] FIGS. 7A-7B collectively provide a flow chart illustration ofprocessing, steps 700 performed by the audio conference mixer 90 (FIG.4) at least once every 125 μseconds to create a summed conference signalfor each conference. The executable program instructions associated withthe processing steps 700 are typically downloaded by the controller/CPU48 (FIG. 2) over the system bus 50 (FIG. 2) to the DSP designated toperform the conference mixer function. The download may occur duringinitialization or system reconfiguration.

[0041] Referring to FIG. 7A, for each of the active/assigned ports ofthe audio conferencing system, step 702 reads the speech bit and theDTMF detect bit received over the TDM bus 42 (FIG. 4). Alternatively,the speech bits may be provided over a dedicated serial link thatinterconnects the audio processor and ;the conference mixer. Step 704 isthen performed to determine if the speech bit for the port is set (i.e.,was energy detected on that port?). If the speech bit is set, then step706 is performed to see if the DTMF detect bit for the port is also set.If the DTMF detect bit is clear, then the audio received by the port isspeech and the audio does not include DTMF tones. As a result, step 708sets the conference bit for that port, otherwise step 709 clears theconference bit associated with the port. Since the audio conferencingplatform 26 (FIG. 1) can support many simultaneous conferences (e.g.,384), the controller/CPU 48 (FIG. 2) keeps track of the conference thateach port is assigned to and provides that information to the DSPperforming the audio conference mixer function. Upon the completion ofstep 708, the conference bit for each port has been updated to indicatethe conference participants whose voice should be included in theconference sum.

[0042] Referring to FIG. 7B, for each of the conferences, step 710 isperformed to decompress each of the audio signals associated withconference bits that are set. Step 711 performs AGC and gain/TLPcompensation on the expanded signals from step 710. Step 712 is thenperformed to sum each of the compensated audio samples to provide asummed conference signal. Since many conference participants may bespeaking at the same time, the system preferably limits the number ofconference participants whose voice is summed to create the conferenceaudio. For example, the system may sum the audio signals from a maximumof three speaking conference participants. Step 714 outputs the summedaudio signal for the conference to the audio processors. In a preferredembodiment, the summed audio signal for each conference is output to theaudio processor(s) over the TDM bus. Since the audio conferencingplatform supports a number of simultaneous conferences, steps 710-714are performed for each of the conferences.

[0043]FIG. 8 is a flow chart illustration of processing steps 800performed by each audio processor to output audio signals over the TDMbus to conference participants. The executable program, instructionsassociated with these processing steps 800 are typically downloaded toeach audio processor by the controller/CPU during system initializationor reconfiguration. These, steps 800 are also preferably executed atleast once every 125 μseconds.

[0044] For each active/assigned port, step 802 retrieves the summedconference signal for the conference that the port is assigned to. Step804 reads the conference bit associated with the port, and step 806tests the bit to determine if audio from the port was used to create thesummed conference signal. If it was, then step 808 removes the gain(e.g., AGC and gain/TLP) compensated audio signal associated with theport from the summed audio signal. This step removes the speaker's ownvoice from the conference audio. If'step 806 determines that audio fromthe port was not used to create the summed conference signal, then step808 is bypassed. To prepare the signal to be output, step 810 applies again, and step 812 compresses the gain corrected signal. Step 814 thenoutputs the compressed signal onto the TDM bus for routing to theconference participant associated with the port, via the NIC (FIG. 2).

[0045] Notably, the audio conferencing platform 26 (FIG. 1) computesconference sums at a central location. This reduces the distributedsumming that would otherwise have to be performed to ensure that theports receive the proper conference audio. In addition, the conferenceplatform is readily expandable by adding additional NICs and/orprocessor boards. That is, the centralized conference mixer architectureallows the audio conferencing platform to be scaled to the user'srequirements.

[0046]FIG. 9 is a flow chart illustration of processing steps 900performed by the transcoder 95 (also referred to as an encoder). Theexecutable program instructions associated with these processing steps900 are typically downloaded to the transcoding circuit by thecontroller/CPU during system initialization or reconfiguration. Thesesteps 900 are also preferably executed at least once every 125 μseconds.

[0047] For each conference that the system is supporting—the transcoder95 (FIG. 4) executes step 902 to read the conference sum associated withthe conference. Step 904 is then performed to transcode the conferencesum signal into a format that is suitable for transmission over theInternet. For example, step 904 may involve transcoding the conferencesum from μ-LAW format to a format that is suitable for streaming theaudio conference onto the Internet in real-time. Step 906 is thenperformed to output the transcoded sum onto the system bus 50. Referringagain to FIG. 2, the transcoded sum is output on the system bus 50 tothe controller/CPU 48, which outputs the transcoded sum on the data link36 to the server 34 (FIG. 1). The server then streams the transcoded sumto conference participants via the Internet/intranet.

[0048] The transcoding may be performed using the REALPLAYER™ streameravailable from Real Networks. In general, the transcoder 95 (FIG. 4)performs the task of streaming audio conferences onto the Internet (andintranets) in real-time. One of ordinary skill in the art will recognizethat transcoding/encoding techniques other than those provided by theREALPLAYER™ real-time streamer may also be used. In addition, thepresent invention is clearly not limited to the preferred embodimentillustrated herein. It is contemplated that the method of streaming areal-time audio conference to conference participants via the Internetmay be performed a number of different ways. For example, rather thanhaving the server physically separate from the audio conferenceplatform, the server function may be integrated into the audioconference platform. In addition, the server may also receiverequests/data over the Internet/intranet such as a question from aparticipant, which can be routed to the other conference participantseither by the server in the form of text over the Internet/intranet, asynthesized voice or the actual voice.

[0049] One of ordinary skill will appreciate that as processor speedscontinue to increase, that the overall system design is a function ofthe processing ability of each DSP. For example, if a sufficiently fastDSP was available, then the functions of the audio conference mixer, theaudio processor and the DTMF tone detection and the other DSP functionsmay be performed by a single DSP.

[0050] Although the present invention has been shown and described withrespect to several preferred embodiments thereof, various changes,omissions and additions to the form and detail thereof, may be madetherein, without departing from the spirit and scope of the invention.

What is claimed is:
 1. An audio conferencing system, comprising: anaudio conference mixer that receives digitized audio signals and sums aplurality of said digitized audio signals containing speech to provide asummed conference signal; and a transcoder that receives and transcodessaid summed conference signal to provide a transcoded summed signal thatis streamed onto the Internet.
 2. An audio conferencing system,comprising: a data bus; a plurality of digital signal processors adaptedto communicate on said data bus, wherein a first of said plurality ofdigital signal processors receives digitized audio signals associatedwith conference participants who are speaking, and sums a plurality ofsaid digitized audio signals to provide to a second of said plurality ofdigital signal processors a summed conference signal and a conferencelist indicative of said digitized audio signals summed to generate saidsummed conference signal, wherein for each conference participant onsaid conference list said second of said plurality of digital signalprocessors removes said digitized audio signal associated with theconference participant from said summed conference signal to provide aunique conference signal for the conference participant; and means forreceiving and transcoding said summed conference signal to provide atranscoded summed signal that is streamed onto the Internet.
 3. Theaudio conferencing system of claim 2, wherein said means for receivingand transcoding comprises a digital signal processor adapted to receivesaid summed conference signal and process said summed conference signalto provide a transcoded summed signal that is streamed on the Internet.4. The audio conferencing system of claim 2, wherein said first of saidplurality of digital signal processors is configured as an audioconference mixer, said second of said plurality of digital signalprocessors is configured as an audio processor that receives saiddigitized audio signals and determines which of said digitized audiosignals comprises voice data and provides a speech list indicativethereof to said audio conference mixer, which sums a plurality of saiddigitized audio signals identified in said speech list to provide saidsummed conference signal.
 5. The audio conferencing system of claim 4,wherein said speech list comprises a plurality of speech bits, eachuniquely associated with one of said digitized audio signals.
 6. Theaudio conferencing system of claim 5, wherein said conference listcomprises a plurality of conference bits, each uniquely associated withone of said digitized audio signals.
 7. The audio conferencing system ofclaim 2, further comprising: a system bus; and a controller thatcommunicates with said plurality of digital signal processors over saidsystem bus, and downloads executable program instructions to saiddigital signal processors.
 8. The audio conferencing system of claim 4wherein said audio processor provides said plurality of digitized audiosignals to said audio conference mixer over a dedicated communicationslink between said audio processor and said audio conference mixer. 9.The audio conferencing system of claim 4 wherein said audio processorprovides said plurality of digitized audio signals to said audioconference mixer over said data bus.
 10. An audio conferencing platform,comprising: means for receiving audio signals associated with conferenceparticipants, and for providing a digitized audio signal and a speechbit for each of said audio signals, wherein said speech bit indicateswhether or not said associated digitized audio signal includes voicedata from the associated conference participant; an audio conferencemixer adapted to receive said digitized audio signals and said speechbits, and sum a plurality of said digitized audio signals based upon thestate of said speech bits to provide a summed conference signal, andprovide a conference list indicative of the conference participantswhose voice is included in said summed conference signal; means forreceiving said summed conference signal and said conference list, forproviding said summed conference signal to each of said conferenceparticipants that are not on said conference list, and for eachconference participant on the conference list removing the digitizedaudio signal associated with that conference participant from saidsummed conference signal and providing a resultant difference audiosignal to the conference participant on said conference list; andcircuitry adapted to transcode said summed conference signal to providea transcoded summed signal that is streamed onto the Internet.
 11. Theaudio conferencing platform of claim 10, wherein said audio conferencemixer comprises a first digital signal processor.
 12. The audioconferencing platform of claim 11, wherein said means for receivingaudio signals comprises a network interface circuit and a second digitalsignal processor configured to operate as an audio processor, whereinsaid network interface circuit and said audio processor areinterconnected by a time division multiplex (TDM) bus.
 13. The audioconferencing platform of claim 10, wherein said means for receiving saidsummed conference signal and said conference list comprises a digitalsignal processor.
 14. The audio conferencing platform of claim 10,further comprising a time division multiplex (TDM) bus thatinterconnects (i) said means for receiving audio signals associated withconference participants, (ii) said audio conference mixer and (iii) saidmeans for receiving said summed conference signal and said conferencelist, wherein said summed conference signal and said conference list areprovided over said TDM bus.
 15. The audio conferencing platform of claim10, wherein said audio conferencing platform supports a plurality ofsimultaneous conferences and said means for receiving audio signalsfurther comprises, means for DTMF tone detection that tests each of saidaudio signals to determine if a DTMF tone is present and provides a DMTFdetect bit indicative thereof, wherein each of said audio signals has auniquely associated DTMF detect bit; and said audio conference mixercomprises means for checking said DTMF detect bit associated with anydigitized audio signal to be added to said summed conference signalbased upon said speech list, to ensure that said summed conferencesignal does not include digitized audio signals whose associated DTMFdetect bit indicates the presence of a DTMF tone.
 16. An audioconferencing platform that provides a summed conference signal over theInternet, said platform comprising: input circuitry adapted to receivedaudio signals associated with conference participants, and provide adigitized audio signal and a speech bit for each of said audio signals,wherein said speech bit indicates whether or not said associateddigitized audio signal includes voice data from the associatedconference participant; a centralized audio conference mixer adapted toreceive said digitized audio signals and said speech bits, and sum aplurality of said digitized audio signals based upon the state of saidspeech bits to provide a summed conference signal, and provide aconference list indicative of the conference participants whose voice isincluded in said summed conference signal; an encoder that receives andtranscodes said summed conference signal to provide a transcoded summedsignal that is streamed onto the Internet.